On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso <a_villa...@palosanto.com> wrote: > As a continuation of my project, I am trying to set up Kamailio as a > Websocket bridge to Asterisk. The asterisk instance is running as localhost, > with its own websocket support disabled, but otherwise has accounts with all > of the avfp and dtls settings for websockets. Additionally, I have removed > the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 > and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to > localhost. This allows asterisk to bypass rtpproxy when signaling through a > websocket. I have already established calls originating from the browser. > However, I have an issue with the registration. > Just in passing, why did you remove bindaddr=127.0.0.1?
> In my setup, Kamailio receives the REGISTER from whatever source, and > forwards this through UDP to Asterisk, after the multiple-domain > transformation. Therefore, Asterisk sees the following in its SIP port (all > traffic through localhost): > > REGISTER sip:pbx.villacis.com SIP/2.0 > Via: SIP/2.0/UDP > 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 > Via: SIP/2.0/WSS > r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 > Max-Forwards: 69 > To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080> > From: "Alex Villac..s" > <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac > Call-ID: vp2akar0aqfmgfa6m1taau > CSeq: 82 REGISTER > Contact: > <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad>";expires=600 > Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE > Supported: path,gruu,outbound > User-Agent: SIP.js/0.6.2 > Content-Length: 0 > > Asterisk answers this through UDP, and Kamailio forwards it through the > websocket: > > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 > Via: SIP/2.0/WSS > r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 > From: "Alex Villac..s" > <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=b5c0lq4kac > To: <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=as5ae2df76 > Call-ID: vp2akar0aqfmgfa6m1taau > CSeq: 82 REGISTER > Server: Asterisk PBX 11.12.0 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Expires: 600 > Contact: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;expires=600 > Date: Thu, 28 Aug 2014 22:21:15 GMT > Content-Length: 0 > > Then Asterisk sends this through UDP, and Kamailio again forwards it through > the websocket: > > NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 > Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport > Max-Forwards: 70 > From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 > To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws> > Contact: <sip:asterisk@127.0.0.1:5080> > Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 > CSeq: 102 NOTIFY > User-Agent: Asterisk PBX 11.12.0 > Event: message-summary > Content-Type: application/simple-message-summary > Content-Length: 89 > > Messages-Waiting: no > Message-Account: sip:*97@127.0.0.1:5080 > Voice-Message: 0/0 (0/0) > > Since I have not implemented handling of voicemail indications, the browser > answers this: > > SIP/2.0 405 Method Not Allowed > Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 > To: <sip:fnuql6ft@192.168.3.2:47307;transport=ws>;tag=ggu5etber9 > From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as43c12840 > Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 > CSeq: 102 NOTIFY > Supported: outbound > Content-Length: 0 > > > After that, Asterisk wants to send an OPTIONS packet. From the point of view > of Asterisk (sip set debug on), it is already sent, but never gets a > response. However, tcpdump shows that the packet is never sent through the > localhost interface in the first place. It is also not sent through any > other interface. My guess is that since the REGISTER has a contact with > transport=ws , Asterisk wants to send this through a websocket (which is > disabled). So I could have to generate a contact without transport=ws . > > I have worked around this by setting qualify=no in the account for the > websocket, but I would like a better solution, one that allows the OPTIONS > packet to reach the browser, and to get the response. What is the proper way > to deal with this? > What does the OPTIONS message in asterisk look like? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
